H.323 Gateway Call Forwarding to Cell Phone Issue

I have a site that is using an H323 GW that is configured to use a CUCM 6.0/Unity 4.0 cluster. There is a 1MB line on the gateway for incoming calls which forwards to an AA on the Unity server. Normal inbound and outbound calls work fine. However, if a user forwards their Cisco IP phone to their cell phone it fails:

The auto attendant tells you âwait while I transfer your callâ and then you hear 3 beeps. The user's cell rings once and then the call disconnects on the user's phone. The call stays connected on the caller's phone.

Replies

I think I've seen something like this before. You need to define a Media Termination Point and associate that with a Media Rersource Group and MRG List, and make sure that the device pool that includes the voicemail ports is assigned that MRGL. At least, I beleive that is the case. It has to do with hairpinning on a H.323 Gateway.

Okay, in CUCM I set the gateway config to use a MRGL and to require a MTP. Now when I call the 1MB and get the AA I can dial the extension, it says "wait while I transfer your call..." then there's about 1-15 seconds of silence. After that a recording says that the extension is not available and to leave a message.

I assume that the calling search space is configured correctly. I doubt you would have received the single ring you were getting before if it were a CSS problem. Did you perhaps modify the cell number within Unity during your troubleshooting to something that isn't valid?

OK, let me take a step back here... You have a 1MB analog line coming into a VIC-FXO port on the H.323 router, yes? You call in on that line, Unity answers, sends the call to the CUCM extension, which is configured to CFAll to a cell phone number... I assume you have some other PSTN access to allow the call to get out to the cell phone, right? Or does the 1MB line support mulitple calls in a trunk; I am under the impression that a 1MB is simply a metered 1FB, which support only a single call.

So, the issue only comes up when an analog call from the PSTN comes into a H323 gateway which connects to an IP-based AA and then gets forwarded to a SIP provider. So the real problem has to do with the analog-to-SIP hairpinning within the gateway. I was assuming we were talking about analog-to-analog hairpin.

Can you see the call toward your SIP provider within the gateway via debugs? I'm not familiar with SIP trunking from H323, so I don't know what debug commands would be best.